I. Field of the Invention
The present invention pertains generally to the field of wireless communications, and more specifically to methods and apparatus for producing non-symmetric links over-the-air in a wireless communication system.
II. Background
Transmission of voice by digital techniques has become widespread, particularly in long distance and digital radio telephone applications. This, in turn, has created interest in determining the least amount of information that can be sent over a channel while maintaining the perceived quality of the reconstructed speech. If speech is transmitted by simply sampling and digitizing, a data rate on the order of sixty-four kilobits per second (kbps) is required to achieve a speech quality of conventional analog telephone. However, through the use of speech analysis, followed by the appropriate coding, transmission, and resynthesis at the receiver, a significant reduction in the data rate can be achieved.
Devices for compressing speech find use in many fields of telecommunications. An exemplary field is wireless communications. The field of wireless communications has many applications including, e.g., cordless telephones, paging, wireless local loops, wireless telephony such as cellular and PCS telephone systems, mobile Internet Protocol (IP) telephony, and satellite communication systems. A particularly important application is wireless telephony for mobile subscribers.
Various over-the-air interfaces have been developed for wireless communication systems including, e.g., frequency division multiple access (FDMA), time division multiple access (TDMA), and code division multiple access (CDMA). In connection therewith, various domestic and international standards have been established including, e.g., Advanced Mobile Phone Service (AMPS), Global System for Mobile Communications (GSM), and Interim Standard 95 (IS-95). An exemplary wireless telephony communication system is a code division multiple access (CDMA) system. The IS-95 standard and its derivatives, IS-95A, ANSI J-STD-008, IS95B, proposed third generation standards IS-95C and IS-2000, etc. (referred to collectively herein as IS-95), are promulgated by the Telecommunication Industry Association (TIA) and other well known standards bodies to specify the use of a CDMA over-the-air interface for cellular or PCS telephony communication systems. Exemplary wireless communication systems configured substantially in accordance with the use of the IS-95 standard are described in U.S. Pat. Nos. 5,103,459 and 4,901,307, which are assigned to the assignee of the present invention and fully incorporated herein by reference.
Devices that employ techniques to compress speech by extracting parameters that relate to a model of human speech generation are called speech coders. A speech coder divides the incoming speech signal into blocks of time, or analysis frames. Speech coders typically comprise an encoder and a decoder. The encoder analyzes the incoming speech frame to extract certain relevant parameters, and then quantizes the parameters into binary representation, i.e., to a set of bits or a binary data packet. The data packets are transmitted over the communication channel to a receiver and a decoder. The decoder processes the data packets, unquantizes them to produce the parameters, and resynthesizes the speech frames using the unquantized parameters.
The function of the speech coder is to compress the digitized speech signal into a low-bit-rate signal by removing all of the natural redundancies inherent in speech. The digital compression is achieved by representing the input speech frame with a set of parameters and employing quantization to represent the parameters with a set of bits. If the input speech frame has a number of bits Ni and the data packet produced by the speech coder has a number of bits No, the compression factor achieved by the speech coder is Cr=Ni/No. The challenge is to retain high voice quality of the decoded speech while achieving the target compression factor. The performance of a speech coder depends on (1) how well the speech model, or the combination of the analysis and synthesis process described above, performs, and (2) how well the parameter quantization process is performed at the target bit rate of No bits per frame. The goal of the speech model is thus to capture the essence of the speech signal, or the target voice quality, with a small set of parameters for each frame.
One effective technique to encode speech is multi-mode coding. An exemplary multi-mode coding technique is described in U.S. application Ser. No. 09/217,341, entitled VARIABLE RATE SPEECH CODING, filed Dec. 21, 1998, assigned to the assignee of the present invention, and fully incorporated herein by reference. Conventional multi-mode coders apply different modes, or encoding-decoding algorithms, to different types of input speech frames. Each mode, or encoding-decoding process, is customized to optimally represent a certain type of speech segment, such as, e.g., voiced speech, unvoiced speech, transition speech (e.g., speech occurring between periods of voiced and unvoiced speech), and background noise (silence, or nonspeech) in the most efficient manner. An external, open-loop mode decision mechanism examines the input speech frame and makes a decision regarding which mode to apply to the frame. The open-loop mode decision is typically performed by extracting a number of parameters from the input frame, evaluating the parameters as to certain temporal and spectral characteristics, and basing a mode decision upon the evaluation.
Presently, there is a strong commercial need to increase the efficiency of transmissions within a wireless communication network. As discussed above, the extraction of speech parameters from speech samples to obtain a high target compression factor Cr is one method for creating an efficient system. However, the efficient packing of speech information into binary data packets does not completely address the present problem of reducing bottlenecks in the over-the-air transmission of data packets from a base station to a remote station. In this specification, base station refers to the hardware with which the remote stations communicate. Cell refers to the hardware or the geographic coverage area, depending on the context in which the term is used. A sector is a partition of a cell. Because a sector of a CDMA system has the attributes of a cell, the teachings described in terms of cells are readily extended to sectors.
In a CDMA system, communications between users are conducted through one or more base stations. A first user on one remote station communicates to a second user on a second remote station by transmitting data on the reverse link to a base station. The base station receives the data and can route the data to another base station. The data is transmitted on the forward link of the same base station, or a second base station, to the second remote station. The forward link refers to transmission from the base station to a remote station and the reverse link refers to transmission from the remote station to a base station. In IS95 and IS-2000 systems, the forward link and the reverse link are allocated separate frequencies.
The forward link comprises a plurality of pilot and traffic channels, wherein each channel is spread by an appropriate Walsh or quasi-orthogonal function. Each channel is then spread by a quadrature pair of pseudonoise (PN) sequences at a fixed chip rate of 1.2288 Mcps. The use of Walsh codes and PN sequences allows a base station to generate multiple forward link CDMA channels. The reverse traffic channels can also comprise multiple channels, as specified by the radio configurations of each individual subscriber network.
Each channel is physically constructed to achieve functionally different purposes. For example, a pilot channel may be simply spread using Walsh code xe2x80x9cWo xe2x80x9d but a synchronization channel is an encoded, interleaved, spread, and modulated spread spectrum signal. The other forward and reverse link channels are also encoded, interleaved, spread, and modulated spread spectrum signals, but manipulated with various values to satisfy various requirements imposed by the appropriate telecommunication standard.
A novel and improved method and apparatus for producing non-symmetric links is presented. In one embodiment, the non-symmetric links are for balancing the capacity of a forward link in a wireless communication system with the capacity of a reverse link in the wireless communication system, comprising the steps of: operating a first speech coder with a first set of modes on the forward link, wherein at least one mode in the first set of modes is associated with a low average data rate; and operating a second speech coder in a second set of modes on the reverse link, wherein each mode in the second set of modes is associated with a high average data rate.
In another embodiment, a method for transmitting frames of data in a forward link and a reverse link of a wireless communication system is presented, the method comprising the steps of: operating a forward link speech coder in a first plurality of modes; and operating a reverse link speech coder in a second plurality of modes, wherein at least one mode in the second plurality of modes is different from each mode in the first plurality of modes.
In another embodiment, a method for transmitting frames of data in a forward link and a reverse link of a wireless communication system is presented, the method comprising the steps of: operating a forward link speech coder in a first plurality of modes; and operating a reverse link speech coder in a second plurality of modes, wherein at least one mode in the first plurality of modes is different from each mode in the second plurality of modes.